[asterisk-users] R: GXP2000 and asterisk 1.0.9
1. The phone has not the DND active, i checked it several times 2. Outbound calls always success, the problem is when the phone receive a call, it repsnds with busy signalling. 3. The firmware i just the lastest one 1.1.5.15 and i cannot upgrade asterisk. Thanks for all -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di C F Inviato: mercoledì 13 febbraio 2008 21.09 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 Just check DND if it's on on the phone or not. What is the CLI output when you try making a phone call? Why don't you try it with a later version of astrisk and a Phone? On Feb 13, 2008 10:58 AM, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all gusy, i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in busy state, if you call it get the busy tone but the phone can male any type of call. This is my sip.conf [502] language = it username = 502 secret = password host = dynamic type = friend context = local canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 callerid = 502 502 Under Grandstream's support suggest, I set Use randmom port to yes and Nat traversal (STUN) to No, but send keep alive but without success. This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6 Anyone can help me ? Thanks in advance Giordano No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 20.00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] R: GXP2000 and asterisk 1.0.9
Thanks Henry, anyway the phone is always registered when i get the busy tone * Name : 502 Secret : Set MD5Secret: Not set Context : local Language : it FromUser : FromDomain : Callgroup: 1 (2) Pickupgroup : 1 (2) Mailbox : LastMsgsSent : -1 Dynamic : Yes Expire : 703 seconds Expiry : 900 Insecure : No Nat : No ACL : No CanReinvite : No PromiscRedir : No DTMFmode : info LastMsg : 0 ToHost : Addr-IP : 192.168.13.171 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Username : 502 Codecs : 0x8010f (g723|gsm|ulaw|alaw|g729|h263) Codec Order : (alaw|ulaw|gsm|g729|g723) Status : OK (22 ms) Useragent: Grandstream GXP2000 1.1.5.15 Full Contact : sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone Any idea? Thanks again to all -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Henry Devito Inviato: mercoledì 13 febbraio 2008 22.01 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 Is your phone actually registered to the switch. go to the CLI and do a 'sip show peers' see if extension 502 is registered. Making an outbound call does not prove that the phone is registered. - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 2:09 PM Subject: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 Just check DND if it's on on the phone or not. What is the CLI output when you try making a phone call? Why don't you try it with a later version of astrisk and a Phone? On Feb 13, 2008 10:58 AM, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all gusy, i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in busy state, if you call it get the busy tone but the phone can male any type of call. This is my sip.conf [502] language = it username = 502 secret = password host = dynamic type = friend context = local canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 callerid = 502 502 Under Grandstream's support suggest, I set Use randmom port to yes and Nat traversal (STUN) to No, but send keep alive but without success. This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6 Anyone can help me ? Thanks in advance Giordano No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 20.00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog DID
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 darren wrote: An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in on a DID the telco picks up the first available line (remember, the customer is providing dial tone.) and dials the last 4 digits of the dialed number. They are often replaced by PRIs but in some locations a PRI is not affordable and these provide the same DID functionality for a small fraction of the price. We've done installs on the same here. Basically you set up FXS connections and the telco picks them up as if they were a telephone, the extensions get dialled and it kinda just works. Only really useful where you can't get a BRI here in New Zealand. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHtAOYDQNt8rg0Kp4RAjq4AJ44bv6pvzyV8jELlOAugHm60cF89QCcCXVi 1V/DGRiH61DV2IWqVZU5MXU= =v6U7 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK issue - Asterisk dialling 999... sort of
In the UK, to make us match the rest of Europe, it's also possible to access the Emergency Services on 112. Again, although a few calls were made around the right time, none of them were 999 or 112. The I've examined the master.csv for 30 mins before the Police said the call was made, and can't see any possible way the number was dialled. We do have a special entry in our call plan to deal with emergency calls, which was taken from voip-info, that intercepts an internal extension of 999, and sends it out via a ZAP channel. We even have it setup so if it fails to find a free ZAP channel, it will end the call on channel 1 and then dial. Again though, this is only for a calls specifically made to 999. I'm baffled... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Langstaff Sent: 13 February 2008 15:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] UK issue - Asterisk dialling 999... sort of It might be possible to get to the emergency service via 112 or a local number as well. Do you have *any* calls placed at about the time of the 999 calls? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Phil Knighton Sent: 13 February 2008 14:12 Hello This is a fun one for the list... Twice now, the Police have contacted us to say they have had a silent call then hangup from our landline number to the 999 service. As a matter of course, they follow up these calls in case someone is in distress. Nobody here was in distress - well, no more than normal! The Police aren't hugely happy when we tell them it must be a mistake. Thing is, I have checked both our master log, and our dialled calls log - and nobody dialled 999! Each phone has an account code applied from sip.conf, and we log all numbers dialled. Nobody dialled out. There are no phones connected in anyway other than via Asterisk, fax number is dealt with by a virtual machine, alarm system is on a different number... Any ideas before the rossers come and take me away? Phil http://www.mjog.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK issue - Asterisk dialling 999... sort of
Hi Tilghman As far as I can see from both master.csv and the account log, no number was dialled beginning 999 (or 112 - both numbers connect to the Emergency Services, and the Police couldn't tell me which had been called). Unfortunately, my Telco (British Telecomsigh) can't tell me exactly when it was called, which line it was on or anything else! All I have is the information the Police gave me. Is it possible for Asterisk (for whatever reason) to pass a call that isn't logged in master.csv? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: 13 February 2008 15:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] UK issue - Asterisk dialling 999... sort of On Wednesday 13 February 2008 08:12:25 Phil Knighton wrote: Thing is, I have checked both our master log, and our dialled calls log - and nobody dialled 999! Each phone has an account code applied from sip.conf, and we log all numbers dialled. Nobody dialled out. Have you checked all numbers that might have a PREFIX of 999? Here in the States, occasionally a prankster will tell someone annoying her to call her on her cell phone at 911-5924 or something like that, and of course, the system only sees the 911 portion, not the additional 4 digits, which connects them to the emergency number on this side of the Pond. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK issue - Asterisk dialling 999... sort of
Can I just say I'm grateful for all the replies - this list is invaluable. Thanks for the suggestion Razza, I've been back again to the logs and no call was placed that contained the string 999 or 112 at the right time! Glad it made you smile, said it was a fun one for the list. Looking like this one is going to go down as a mystery... until the Police call again! It has only happened twice in the 16 months we've been using Asterisk, I'm just keen to find a solution. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Razza Sent: 13 February 2008 21:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] UK issue - Asterisk dialling 999... sort of When I first set up asterisk, I had similar, fortunately not with the old bill! It basically boiled down to not enough delay between seizing the line and dialing the digits, for example the following would have dialled the coppers 01299 912345, as 012 would have been missed. I'm guessing this isn't whats happening to you, if all your other calls are uworking fine, but did bring back some memories and made me smile :o) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple host in 1 context on sip.conf
Hi Mark! 13 feb 2008 kl. 23.42 skrev Mark Quitoriano: Is it possilble for a single context to have multiple host= something like this First context is something we use to describe a segment of the dialplan. I would call this section. [carrier] host=ip address1 host=ip address2 host=ip address3 type=peer disallow=all allow=g729 allow=ulaw canreinvite=no insecure=yes qualify=yes No. You can only add one. Normally I would add host=sip.mydomain.com and have multiple DNS entries or use SRV records to do failover and such, provided you use this for outbound calls. Since you call this peer carrier I assume you want to handle inbound calls. Today, you will have to define three different peers, but remember that you can use templates. [carrier](!) type=peer disallow=all allow=g729 allow=ulaw canreinvite=no insecure=yes qualify=yes [carrier-01](carrier) host=ip address1 [carrier-02](carrier) host=ip address2 [carrier-03](carrier) host=ip address3 You will now have three peers named carrier-01-03 but no peer named carrier in your sip driver when you run sip show peers. Regards, /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream GXP2000 Loses Connectivity
I've had the opposite problem. Press mute while the call is still ringing and it will say MUTE on the display but the microphone is not muted. It was very embarrassing to discover this bug. On Wed, Feb 13, 2008 at 2:03 AM, Thomas Kenyon [EMAIL PROTECTED] wrote: Lutgring, Sam wrote: I take it you've also not had the problem where the handset microphone stops working. (This is apparently already fixed and will be available in the next beta firmware release (1.1.6.x), when they've fixed some more problems that have been very difficult to track down.) I'd like to go back to 1.1.5.15 if nothing more than for the improved audio quality and the on-hook dialling. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK -999 dialing issue
Hi Amit
OK, the majority of our calls go out via zaptel fxo and pstn lines.
When these are all busy, calls are routed via a VOIP provider here in
the UK. All activity is recorded in our logs, and I can find no trace
of either 999 or 112 (if since been reminded that in the UK, you can now
also use 112 which is consistent with continental Europe).
I can't find a call placed at the relevant time that had these numbers,
even as mid-part of a string.
Below is the part which deals with our external calls. As you can see,
calls are routed out via zap, or VOIP (that's the gradwell bit). If
someone prefixes a call with 9 it forces it our via VOIP and if
someone dials 999 it is intercepted and sent via the zap channels. If
no zap channel is free, a call on channel 1 is ended and the number
re-dialled. This makes sure that emergency calls can always be placed
on a landline.
Any ideas would be appreciated!
Phil
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten = _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
; The below section will allow for 3 digit BT numbers to be called, by
prefixing them with 9
; For example: 154 is BT Business Faults - dial 9154
exten = _9[123456789]XX,1,NoOp(${EXTEN})
exten = _9[123456789]XX,2,Dial(Zap/g0/${EXTEN:1},,j)
; The below section will allow for 999 Emergency calls to be made. This
will FORCE these calls
; over our BT lines, which will provide CallerID and location
information to the Emergency Operator
; If there are no BT lines free, it will force a call to end and then
dial
exten = 999,1,NoOp(999)
exten = 999,2,Dial(Zap/g0/999,,j)
exten = 999,3,Hangup()
exten = 999,102,SoftHangup(Zap/1-1)
exten = 999,103,Wait(1)
exten = 999,104,Goto(1)
[softoption-gradwell]
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten = _0[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _[1-9]X,1,Dial(IAX2/Gradwell/441353${EXTEN},,)
From: amit salunkhe [mailto:[EMAIL PROTECTED]
Sent: 14 February 2008 07:44
To: Phil Knighton
Subject: UK -999 dialing issue
HI Phil
Can u send me ur out call context config. Also tell me what
ur using with Asterisk to make out call SIp-Voip or Pstn line with Fxo
card?
also check with this command in ur Asterisk console. sip
show peers so u can get anybody from out side place such call inbehalf
of u. check who how many user regsiter with ur Asterisk. if ur using
FXO card then also there is chance to check this.
also use Mysql for CDR table tocheck who try to call at time. so
u got any hint for this
Regards
Amit
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Re: [asterisk-users] PCI32 and PCI-X compatibility
Thanks Michael, that's a *huge* thing you're telling me, in the wiki details for the PCI-X bus I've read about retrocompatibility, but I just wanted to be 100% sure. I can go on and order my server, now! Thanks again Marco ps. This proves also the complete unaccuracy of the information provided by the local Digium distributor - italian people, be aware! Michael Spiceland ha scritto Marco, You should not have any issues using a PCI card in a PCI-X slot, as long as the card is a 3.3V PCI card. The cards that you mention above are 3.3v compatible and you should be able to use them. All of Digium's product line is available for 3.3v slots. Most are universal and can be used in 3.3v or 5v slots. The only exceptions are the dual and quad span T1/E1 digital cards. For those cards, there are 3.3v variants (TE410P and TE210P) and 5v variants (TE405P and TE410P). Oops, I meant that the 5v variants are the TE405P and *TE205P*. 3.3v - TE410P and TE210P 5v - TE405P and TE205P Michael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK -999 dialing issue
On Thu, 14 Feb 2008, Phil Knighton wrote:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten = _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
OMG!!!
You're selecting 2 different output channels depending on the number
dialled!!!
(UK or international)...
That's ... LCR!!!
In ... Dialplan!!!
And according to a recent thread, that's like ... impossible, not
recommended, really really hard, with databases and external hardware
required, etc. (!!!)
(sorry)
Gordon
(dialplan junkie)
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Re: [asterisk-users] MWI problem with Siemens Gigaset S675 IP
The 481 Call Leg/Transaction Does Not Exist response to the NOTIFY makes me think that you might need to configure the phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages from the phone when it is booted? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaap Winius Sent: 13 February 2008 18:46 Hi list, Before purchasing a number of Siemens DECT SIP phones, the Gigaset S675 IP, I read that the problems with MWI had been fixed with the latest firmware version (see http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm not so sure that's the case. After setting up a network mailbox for one of these phones, as well as an Asterisk voicemail account (ext. 1000) in voicemail.conf's default context, I added the following line to my phone's context in sip.conf: mailbox=1000 However, soon after executing a 'sip reload' on the console, the following error message will appear every three minutes: [Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. The IP address belongs to my server. At the same time, I used tcpdump to see what else might be going on. I found the following: 19:18:22.540113 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 545 [EMAIL PROTECTED] .)..NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0 19:18:22.571452 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 308 E..P...f... .a_SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: The latest comment on the voip-info.org page above outlines the same problem. Can anyone here confirm that this is indeed a Siemens problem, or might it be an Asterisk problem after all? I'm running Asterisk v1.4.14 on a Debian etch server. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ser, asterisk and ip2ipgw
Hi, i use a ser, as proxy sip server(authentication), then a cisco router as sip2h323 gw(authorization and accounting). i want to start asterisk as sip statefull b2bua server, any suggestion to howto or documentation to asterisk integration and b2b use? ty in advance. -- Riccardo Cupardo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK -999 dialing issue
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten = _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
Just an aside -
1) For clarity, could you use 'Z' here instead of '[123456789]'?
2) It does not look like you would be able to dial numbers that
start with 0[123456789] and then have subsequent zeros
(e.g. 01xx xxx ) - is that your intent?
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[asterisk-users] Error checking asterisk method - suggestions?
Hi there dear users and dear developers of Asterisk!
I've got a maybe strange idea, let's see if you laugh or think it's reasonable J
I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering
machine).
Each analog line can be reached through a phonenumber, since they are each
connected to my telephone provider. Yes expensive I know J
The challenge:
I'd like to somehow verify that my 1) TDM800P cards and 2) the analog lines,
and 3) my operator is alive and working, and I have an Idea which I wonder if
will/could work.
My first idea was to ask the zap-driver if it could detect if the line was ok,
but no function existed to do that, what I could find. Anyone knows about one?
My second idea, was to try calling simply, to know if things were ok. But, I
couldn't just call any number, I had to know the number was in use, and not
disturbing anyone.
So, I called myself, or I called another of my phonelines.
So,
I'd like to use the asterisk manager interface in java to originate a call from
one ZAP-channel, calling out to my telephone provider,
And then they will direct the call back to my, but into another ZAP-channel
(since I'm calling that channel's number).
So: I'm making ZAP/1 calling out to no 323121321 - telephone company, Ok:
323121321 belongs to this guy - redirecting me to my ZAP/2 channel, which
answers the call.
Then I have a connection, and ZAP/2 will answer and do some DTMF.
My first ZAP/1 is run through my java program, and I'd like to listen for
certain DTMF-tones, to know I have a working circuit.
The goal for all of this, is to verify things are working, so my provider is
not down, or one of my ZAP-lines are dead.
So far, I've tried calling and got some half-success, but not sure what is
going on doing all the right way.
For ex: why am I calling with Zap1, to Zap 3, and then Zap 7 is answering? 3
channels used for one outgoing and one incoming call? Something must be very
wrong J
Please educate me, dear experts.
Input?
Sincerely,
Johan Sandgren
www.svep.sehttp://www.svep.se, [EMAIL PROTECTED]
Frosty Sweden but with some sunshine today !! J
My code and settings below, for information.
=JAVA CODE (extract)
OriginateAction originateAction = new OriginateAction();
ManagerResponse originateResponse = null;
originateAction.setChannel(ZAP/1);
originateAction.setContext(Outgoing);
originateAction.setExten(201); // maps to
ZAP/7 through external phonecompany PBX
originateAction.setPriority(new Integer(1));
originateAction.setTimeout(new Long(15*1000));
// xml-milliseconds
originateAction.setAsync(false);
== extensions.conf (extract)
[Incoming]
exten = s,1,Set(DYNAMIC_FEATURES=hangup)
exten = s,2,Agi(agi://localhost/answer.agi)
[Outgoing]
exten = _X.,1,Set(DYNAMIC_FEATURES=hangupfeature)
exten = _X.,n,Dial(Zap/3/${EXTEN}
== Asterisk response:
AGI Debugging Enabled
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'stt' logged on from 127.0.0.1
Channel Zap/1-1 was answered.
-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1,
DYNAMIC_FEATURES=hangupfeature) in new stack
-- Executing [EMAIL PROTECTED]:2] Dial(Zap/1-1, Zap/3/201) in new stack
-- Called 3/201
-- Starting simple switch on 'Zap/7-1'
-- Zap/3-1 answered Zap/1-1
[Feb 14 13:02:33] WARNING[26260]: chan_zap.c:6499 ss_thread: CallerID returned
with error on channel 'Zap/7-1'
-- Executing [EMAIL PROTECTED]:1] Set(Zap/7-1, DYNAMIC_FEATURES=hangup)
in new stack
-- Executing [EMAIL PROTECTED]:2] AGI(Zap/7-1,
agi://localhost/answer.agi) in new stack
AGI Tx agi_network: yes
AGI Tx agi_network_script: answer.agi
AGI Tx agi_request: agi://localhost/answer.agi
AGI Tx agi_channel: Zap/7-1
AGI Tx agi_language: en
AGI Tx agi_type: Zap
AGI Tx agi_uniqueid: 1202990552.2
AGI Tx agi_callerid: unknown
AGI Tx agi_calleridname: unknown
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: unknown
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: Incoming
AGI Tx agi_extension: s
AGI Tx agi_priority: 2
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx
AGI Rx ANSWER
AGI Tx 200 result=0
AGI Rx CHANNEL STATUS
AGI Tx 200 result=6
AGI Rx WAIT FOR DIGIT 1
== Manager 'testmanager' logged off from 127.0.0.1
AGI Tx 200 result=0
AGI Rx CHANNEL STATUS
AGI Tx 200 result=6
AGI Rx WAIT FOR DIGIT 1
AGI Tx 200 result=0
== Spawn extension (Incoming, s, 2) exited non-zero on 'Zap/7-1'
-- Hungup 'Zap/7-1'
= Asterisk log
__
Johan Sandgren
Svep Design Center AB (www.svep.sehttp://www.svep.se)
St. Lars väg 42A, SE-222 70 Lund
Phone: 046-19 27 22
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Re: [asterisk-users] Error checking asterisk method - suggestions?
On Thu, Feb 14, 2008 at 01:17:45PM +0100, Johan Sandgren wrote: Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd like to somehow verify that my 1) TDM800P cards and 2) the analog lines, and 3) my operator is alive and working, and I have an Idea which I wonder if will/could work. My first idea was to ask the zap-driver if it could detect if the line was ok, but no function existed to do that, what I could find. Anyone knows about one? What do you mean by line is OK? My second idea, was to try calling simply, to know if things were ok. But, I couldn't just call any number, I had to know the number was in use, and not disturbing anyone. So, I called myself, or I called another of my phonelines. And you assume noone else calls in at the time? So, I'd like to use the asterisk manager interface in java to originate a call from one ZAP-channel, calling out to my telephone provider, And then they will direct the call back to my, but into another ZAP-channel (since I'm calling that channel's number). For a basic test that the line works, try TestClient and TestServer . Originate a call from testclient (and set there the number. And set all incoming calls temporarily to go to TestServer (did I mention the assumption that noone calls in?) One relatively cheap method of temporary is through setting a global variable to a non-standard value. This means that the non-default value will never last after a reload. And you can set the global even through 'core set global VARNAME VALUE' in the CLI. Check the resulting reports in /var/lib/asterisk/testresults . Make sure all of them were generated, and that none FAIL-ed. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: GXP2000 and asterisk 1.0.9
Try switching your DTMF mode on both asterisk and the phone to RFC2833. I have not seen the issue that you are describing, but I had some very strange issues like call hang-ups when I was using INFO. After switching the issues were gone and I have had no further troubles. Hope this helps you. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis Sent: Thursday, February 14, 2008 3:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9 Thanks Henry, anyway the phone is always registered when i get the busy tone * Name : 502 Secret : Set MD5Secret: Not set Context : local Language : it FromUser : FromDomain : Callgroup: 1 (2) Pickupgroup : 1 (2) Mailbox : LastMsgsSent : -1 Dynamic : Yes Expire : 703 seconds Expiry : 900 Insecure : No Nat : No ACL : No CanReinvite : No PromiscRedir : No DTMFmode : info LastMsg : 0 ToHost : Addr-IP : 192.168.13.171 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Username : 502 Codecs : 0x8010f (g723|gsm|ulaw|alaw|g729|h263) Codec Order : (alaw|ulaw|gsm|g729|g723) Status : OK (22 ms) Useragent: Grandstream GXP2000 1.1.5.15 Full Contact : sip:[EMAIL PROTECTED]:5060;transport=udp;user=phone Any idea? Thanks again to all -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Henry Devito Inviato: mercoledì 13 febbraio 2008 22.01 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 Is your phone actually registered to the switch. go to the CLI and do a 'sip show peers' see if extension 502 is registered. Making an outbound call does not prove that the phone is registered. - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 2:09 PM Subject: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 Just check DND if it's on on the phone or not. What is the CLI output when you try making a phone call? Why don't you try it with a later version of astrisk and a Phone? On Feb 13, 2008 10:58 AM, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all gusy, i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in busy state, if you call it get the busy tone but the phone can male any type of call. This is my sip.conf [502] language = it username = 502 secret = password host = dynamic type = friend context = local canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 callerid = 502 502 Under Grandstream's support suggest, I set Use randmom port to yes and Nat traversal (STUN) to No, but send keep alive but without success. This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6 Anyone can help me ? Thanks in advance Giordano No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 20.00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK -999 dialing issue
Steve Langstaff wrote:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten = _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
Just an aside -
1) For clarity, could you use 'Z' here instead of '[123456789]'?
2) It does not look like you would be able to dial numbers that
start with 0[123456789] and then have subsequent zeros
(e.g. 01xx xxx ) - is that your intent?
Very good point, he probably wants.
exten = _90ZX.,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
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Re: [asterisk-users] [SPAM] - Re: Error checking asterisk method - suggestions? - Email found in subject
Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd like to somehow verify that my 1) TDM800P cards and 2) the analog lines, and 3) my operator is alive and working, and I have an Idea which I wonder if will/could work. My first idea was to ask the zap-driver if it could detect if the line was ok, but no function existed to do that, what I could find. Anyone knows about one? What do you mean by line is OK? I mean, (being not so educated in the telephone technology), but to know that there is DC voltage connected to my ZAP-channel. (indicating status = OK) According to Wiki, A calling party wishing to speak to another telephone will pick up the handset, thus operating the switch hook, which puts the telephone into active state or off hook with a resistance short across the wires, causing current to flow. So, I suppose to know there is current flowing, would say I'm connected, but probably not guarantee that I can make calls. So this test would not give me trustworthy results. Or what do you say Tzafrir Cohen? (or others :) My second idea, was to try calling simply, to know if things were ok. But, I couldn't just call any number, I had to know the number was in use, and not disturbing anyone. So, I called myself, or I called another of my phonelines. And you assume noone else calls in at the time? Yes, since I only test lines, that haven't had any incoming calls for some time. Sure, someone COULD be using the line right when I'm trying to use it. How would I know it was busy through AMI? Is it possible? So, I'd like to use the asterisk manager interface in java to originate a call from one ZAP-channel, calling out to my telephone provider, And then they will direct the call back to my, but into another ZAP-channel (since I'm calling that channel's number). For a basic test that the line works, try TestClient and TestServer . Originate a call from testclient (and set there the number. And set all incoming calls temporarily to go to TestServer (did I mention the assumption that noone calls in?) One relatively cheap method of temporary is through setting a global variable to a non-standard value. This means that the non-default value will never last after a reload. And you can set the global even through 'core set global VARNAME VALUE' in the CLI. Check the resulting reports in /var/lib/asterisk/testresults . Make sure all of them were generated, and that none FAIL-ed. I will look into this testserver testclient. Totally new stuff to me :) Hope I get it working. Thanks for the tip Tzafrir ! Nice of you! -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue
Hi Tim, Imagine the scenario where we had 10x Asterisk servers, with calls presenting sequentially starting from the first server, then server two, etc. If we took down the first server for maintenance with 'asterisk -rx stop gracefully' we then will block all incoming calls to all servers as our telco will simply relay the BUSY back to the caller. If there are a number of calls on the first server that continue for another 20 minutes, then all inbounds are blocked for that period of time. We are finding at present we have to look at the calls on the server and make a decision if we are busy to simply reboot the server and hence lose calls. Not ideal but then we don't end up blocking our inbounds. What I was hoping to do was find a way to cause the telco to present the call to the next ISDN30 and therefore would allow us to cleanly take down an Asterisk server for maintenance without causing this issue. In a sense to put the ISDN30 into alarm mode while still continuing the active calls. Do you know if this is at all possible, even if we considered patching zaptel to add this functionality or does the telco rely on the entire PRI being in alarm before it presents the call to the next ISDN30 ? This would allow us to run maintenance on our servers during busy periods without causing disruption, and would be an excellent feature. Many thanks, Andrew _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Nelson Sent: 13 February 2008 18:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Even if * is shutdown, zaptel is still running and your ISDN channels are still technically up. Shutting down zaptel should close the channels and put those circuits into alarm mode. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Andrew Smith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 12:03:51 PM (GMT-0600) America/Chicago Subject: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Hi there, I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN E1s. Basically our telco is presenting calls in order of the ISDNs on our servers. SERVER1=1,2,3,4 SERVER2=5,6,7,8 We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in alarm and the calls will then presented to PRIs 5,6,7,8 on SERVER2. If I have to take SERVER1 offline for maintenance (asterisk -rx shutdown gracefully) any incoming calls receive a BUSY tone. What I would like to know is if there is anyway to get around this and not send a BUSY back to our callers and somehow allow our telco to present calls immediately to SERVER2. Anyone have any ideas or are we stuck with this behaviour until the calls drop to 0 and Asterisk shuts down ? Thanks, Andrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK -999 dialing issue
Gordon Henderson wrote:
On Thu, 14 Feb 2008, Phil Knighton wrote:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten = _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
OMG!!!
You're selecting 2 different output channels depending on the number
dialled!!!
(UK or international)...
That's ... LCR!!!
In ... Dialplan!!!
And according to a recent thread, that's like ... impossible, not
recommended, really really hard, with databases and external hardware
required, etc. (!!!)
(sorry)
Gordon
(dialplan junkie)
Not impossible. I think the explanation was that it was ugly. And...
well... that is. Now, imagine sorting through a list of 500,000
possible dialing prefixes (something we have) instead of 3 or 4. Tell me
that would be clean and pretty without a DB lookup.
Anyone can LCR 2 routes in a dialplan, but that's hardly an effective
example of LCR.
N.
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Re: [asterisk-users] UK -999 dialing issue
Steve Langstaff [EMAIL PROTECTED] writes:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten = _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
[..]
2) It does not look like you would be able to dial numbers that
start with 0[123456789] and then have subsequent zeros
(e.g. 01xx xxx ) - is that your intent?
. does not repeat the previous pattern, it simply matches one or more
of anything. _0Z. will happily match 010.
/Benny
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Re: [asterisk-users] [Linux/Python 2.4.2] Forking Python doesn't work
On Wed, 13 Feb 2008 22:26:16 -0500, Russell Bryant [EMAIL PROTECTED] wrote: The arguments to System() are a bit different. Put it in just like you would type at the command line. System(/tmp/netcid.py 2000 Joe) That did it :-) Thanks guys. BTW, for those interested, I didn't have to double-fork: == #!/usr/bin/python import socket,sys,time,os sys.stdout = open(os.devnull, 'w') if os.fork(): sys.exit(0) else: #Here, send broadcast == ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK -999 dialing issue
On Thursday 14 February 2008 03:39:33 Phil Knighton wrote: OK, the majority of our calls go out via zaptel fxo and pstn lines. When these are all busy, calls are routed via a VOIP provider here in the UK. All activity is recorded in our logs, and I can find no trace of either 999 or 112 (if since been reminded that in the UK, you can now also use 112 which is consistent with continental Europe). I can't find a call placed at the relevant time that had these numbers, even as mid-part of a string. I had a recent run-in with the provider who provides my toll-free numbers, as they had gotten a subpoena for the identity of the customer who ran some toll-frees that were being used for fraudulent purposes. It turns out that they had two number transposed prior to getting the subpoena, so not only did they have the wrong customer, they subpoenaed the wrong provider. Consider that if the police will not provide you records of the call, they may have already discovered that they queried the wrong provider. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK -999 dialing issue
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Benny Amorsen
Sent: 14 February 2008 13:57
Steve Langstaff [EMAIL PROTECTED] writes:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN}) exten =
_0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten = _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
[..]
2) It does not look like you would be able to dial numbers that
start with 0[123456789] and then have subsequent zeros
(e.g. 01xx xxx ) - is that your intent?
. does not repeat the previous pattern, it simply matches one
or more of anything. _0Z. will happily match 010.
Oops! Yes, I see that now - my fault for confusing Asterisk pattern
matching with RFC3435 pattern matching. Sorry.
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Re: [asterisk-users] Asterisk Manager and Visual Basic
Bill Andersen wrote: Has anyone tried to used VB6 to communicate with the Asterisk Manager? If so, would you be willing to share some basic code showing your approach to getting connected and parsing results? I've got a Telnet control that is allowing me to connect, authenticate and see the flow of status, etc., but I'm sure there is a better way to do this without using Telnet (maybe not?). Any suggestions? I want to write a presence monitor (a virtual sidecar if you will) Bill As Razza said, you can just use the winsock control included with VB. The protocol is very simple, basically just name/value pairs delimited by #13#10 (CRLF) with an extra CRLF at the end to denote termination of the packet. Action: Originate Channel: local/[EMAIL PROTECTED] Context: to_meetme Exten: s Priority: 1 Variable: CALLERID(num)=123432|CALLERID(name)=Automated Call Async: true extra CRLF == extra CRLF here. So, like this: 1. Send your properly formatted packet to AMI . 2. Read incoming response terminated by double #13#10. 3. Parse values as you are comfortable with. I am in the process of writing a similar product for one of our customers. Well, a re-write to add features and make it cross platform. Here's a screenshot running on Linux/GTK: http://leebo.dreamhosters.com/images/guiApp.png A couple of side notes from what I've learned myself and read on this mailing list or through the wiki: 1. Packet Volume The volume of messages that you can get from the AMI is impressive. I've tested on our Asterisk system which has only 2 pots lines and two sip trunks with 10 desktop phones and the amount of messages can be staggering! Use a proxy for AMI if you have any decent phone traffic. AstManProxy is VERY propular. I wrote one as well, but its still beta and I think there's another one out there somewhere. Usually with these proxy servers you can filter out unwanted/extraneous events to reduce the amount of messages your app has to contend with. 2. Make good use of Observer/Mediator pattern to distribute events to different parts of your GUI. Monolithic loops to write everything out on a timer's event or after a Sleep() for instance, is not a good way to go in my experience. 3. Check the source for manager interface for changes between Asterisk 1.2 and 1.4 (and 1.6?) if you're using 1.2 or plan to. I believe the latest version of AMI is 1.1 (someone can correct me here). A few label names for some of the AMI packets have been changed and a couple events (like LINK event) have been changed drastically. I originally wrote against the 1.2 Manager interface only to find that I had to refactor some code and write descendant classes to handle the slight differences between the two versions' events. I could have saved myself some work had I thought to look for the changes. I think this link is up to date: http://svn.digium.com/view/asterisk/trunk/doc/manager_1_1.txt?revision=98152view=markup Happy coding. -- Warm Regards, Lee Everything I needed to learn in life, I learned selling encyclopedias door to door. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] R: GXP2000 and asterisk 1.0.9
I had GXP-2000's running on 1.0 versions of asterisk even earlier. So I know it does work. I upgraded one of my customers GXP's to the latest firmware in it still works. Can you post the output of the CLI with verbose set to 99 and the the output from the asterisk log file that has the call in it. You can usually do a 'tail /var/log/asterisk/full -n 400' right after the call fails. I will be glad to help, just need a little more info to narrow down the issue. Thanks Henry - Original Message - From: Giordano Grandis [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 14, 2008 2:15 AM Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9 1. The phone has not the DND active, i checked it several times 2. Outbound calls always success, the problem is when the phone receive a call, it repsnds with busy signalling. 3. The firmware i just the lastest one 1.1.5.15 and i cannot upgrade asterisk. Thanks for all -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di C F Inviato: mercoledì 13 febbraio 2008 21.09 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 Just check DND if it's on on the phone or not. What is the CLI output when you try making a phone call? Why don't you try it with a later version of astrisk and a Phone? On Feb 13, 2008 10:58 AM, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all gusy, i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in busy state, if you call it get the busy tone but the phone can male any type of call. This is my sip.conf [502] language = it username = 502 secret = password host = dynamic type = friend context = local canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 callerid = 502 502 Under Grandstream's support suggest, I set Use randmom port to yes and Nat traversal (STUN) to No, but send keep alive but without success. This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6 Anyone can help me ? Thanks in advance Giordano No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 20.00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor Asterisk
Thank you to all those who replied to my last query. For them and for the suggestion, I can monitor asterisk using the asterisk -r -x command option. What I would like to know is that using asterisk -r -x way I can only use the *CLI commands. Is there any other way in which I can monitor asterisk? Moreover it will be very helpful is someone can provide me the C file of the ASTERISK-MIB. I will be using C language to develop my agent for the monitoring of asterisk. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor Asterisk
Thank you to all those who replied to my last query. For them and for the suggestion, I can monitor asterisk using the asterisk -r -x command option. What I would like to know is that using asterisk -r -x way I can only use the *CLI commands. Is there any other way in which I can monitor asterisk? Moreover it will be very helpful is someone can provide me the C file of the ASTERISK-MIB. I will be using C language to develop my agent for the monitoring of asterisk. Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI problem with Siemens Gigaset S675 IP
Quoting Steve Langstaff [EMAIL PROTECTED]: The 481 Call Leg/Transaction Does Not Exist response to the NOTIFY makes me think that you might need to configure the phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages from the phone when it is booted? Yeah, sure. And there are some error messages mixed in too: == 14:01:23.425955 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 473 ... SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:23.426075 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 509 [EMAIL PROTECTED] ...vSIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.5 14:01:23.480238 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 634 E..k... ..F.SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:23.480375 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 432 [EMAIL PROTECTED] ...)SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.10.5:50 14:01:23.918830 arp who-has gigaset.umrk.to tell bitis.umrk.to ../.E .. 14:01:23.921726 arp reply gigaset.umrk.to is-at 00:01:e3:77:f8:67 (oui Unknown) ...w.g../.E .. 14:01:24.539636 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 476 E.. ..2gSUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:24.539816 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 512 [EMAIL PROTECTED] ...ySIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.10.5 14:01:24.594442 IP gigaset.umrk.to.sip bitis.umrk.to.sip: SIP, length: 634 E..i... SUBSCRIBE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 1 14:01:24.594557 IP bitis.umrk.to.sip gigaset.umrk.to.sip: SIP, length: 432 E...- [EMAIL PROTECTED] ...)SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.10.5:50 == Before this was a series of REGISTER messages, and afterwards a series of OPTIONS messages. However, no errors there. Also, this is without having set 'mailbox=1000' or '[EMAIL PROTECTED]' in /etc/asterisk/sip.conf. And, now that I look at it again, the network mailbox settings for the Siemens phone won't have anything to do with these errors either, since it simply makes it possible to associate a button on each handset with an extension used to access a voicemail account. Thanks, Jaap ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] X100P Burnouts
Thought I would post this experience to the list so it's archived for posterity... My company is deploying Asterisk-based PBX's to all of our branch offices. Each office has 2 analog Voice lines and a fax line. We didn't want to go to the expense of using TDM400's in the servers (which run asterisk and Hylafax) so we opted for 2 X100P cards in each box. So far they have worked fine at all but one office. The system at the office in question would work perfectly for an entire day once it was set up. The next morning, however, one of the 2 phone lines would appear to be dead and the X100P would be in Red alarm. Plugging a phone into the X100P's pass-through connection would show dial-tone on the line, and the phone worked perfectly. It was as if the X100p lost it's ability to see the audio on the line and nothing would revive it. Tried restarting the zaptel module, rebooting the server completely, complete power down, unplugging the phone line and even connecting up a phone line simulator and moving the card to another server. The card never works again. This went on for three days. Burned out an X100p every night. I called the telco (Verizon) and they sent out a couple of guys to run tests on the line, but found nothing. Their Demarc is properly grounded and has surge suppression modules attached, the cable that runs from their demarc to our punch-down block is in grounded metal conduit and does not run near any power source. The cable that runs from the punch-down block to the wall jack also does not run anywhere near anything electrical, and everything is twisted pair all the way from the wall jack to the demarc. To further eliminate the possibility of echo or other noise, I ran twisted pair carrying both voice lines from the wall jack to the server, approximately 3 feet in length. Now here is where things get interesting... That 3 foot cable run passes behind a 21 monitor that was connected to the server. When the line tests showed everything OK, I decided the monitor might be a long shot but I could understand how the degaussing coil coil could possibly induce a surge on the phone line if the monitor was somehow degaussing nightly, so I unplugged the monitor's power cable and left everything else as it was. So far so good. X100p #4 is still working this morning, so it looks like the problem is solved. Hope this helps someone else later on. Thanks, Brent Davidson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Asterisk
Soumya Kat wrote: Thank you to all those who replied to my last query. For them and for the suggestion, I can monitor asterisk using the asterisk -r -x command option. What I would like to know is that using asterisk -r -x way I can only use the *CLI commands. Is there any other way in which I can monitor asterisk? Soumya, Yes, asterisk -rx will only allow you to execute CLI commands. It also tends to spew out a bunch of garbage that makes parsing difficult unless verbosity is always set to 0. I recommend taking a look at the Asterisk Manager Interface (AMI) http://www.voip-info.org/wiki-Asterisk+manager+API. It's a cleaner interface that will allow you to read events and issue commands. All of the CLI commands are available through the AMI, as well as an array of additional manager actions. I recently wrote a program that maps the SIP call IDs of the two legs that make up a call using the POE::Component::Client::Asterisk::Manager Perl module http://search.cpan.org/~xantus/POE-Component-Client-Asterisk-Manager/. It provides a simple interface for filtering events, so if you're familiar with Perl I recommend taking a look at it. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ser, asterisk and ip2ipgw
Riccardo Cupardo wrote: Hi, i use a ser, as proxy sip server(authentication), then a cisco router as sip2h323 gw(authorization and accounting). i want to start asterisk as sip statefull b2bua server, any suggestion to howto or documentation to asterisk integration and b2b use? Well, Asterisk is a B2BUA. And it keeps state. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX load balancing
Hello, I've seen that many solutions concerning asterisk dimensioning and load balancing involve the use of sip proxy like openser. Is there any recommended way to balance IAX load? BRs, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] translating iax2 register into sip register
Hello, reading iax2 draft, I'm not sure if the protocol supports peer 2 peer calls (e.g. like SIP). If it doesn't, is Asterisk the only server side iax2 implementation? I also would like to understand if it's possible for asterisk (by means of some configuration rules) to translate a iax2 register into a sip register in order to have user credentials verified by an external entity. I don't like the idea to have all users provisioned within asterisk in order to make them iax capable. Regards, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Manager and Visual Basic
I don't know if it would be of any use to you but we have some C# code that handles the basics of communicating the the Asterisk Manager Interface. It doesn't do anything fancy just sends single commands and checks the responses. We don't use it for monitoring. Regards, Greyman. Thanks for the offer, I think I've got it figured out using winsock. Thanks again. Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pass arguments from extensions.conf
Hi, I have been working with asterisk to make ivr calls (outbound and inbound). I have the functionality - Read(variable|file_name) used in my dialplan. Now i need to pass the variable to my ruby file to compare the data entered with the database (mysql). How can i pass the arguments from my dialplan to the ruby file. Is there a way i can do it with the agi script? Any one has any clues on it. Regards, Naveen.Palani “Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO‑9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI to customers in BFSI, Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global Delivery Model.” This e-mail and any attached files are confidential, proprietary, and may also be legally privileged information, and are intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient of this e-mail, please send it back to the person who sent it to you and delete the e-mail and any attached files and destroy any copies of it; you may call us immediately at + 91 22 2829 0100 or email us at [EMAIL PROTECTED] Quinnox Consultancy Services and/or any of its sister companies owns no responsibility for the views presented in the e-mail and any attached files unless the sender mentions so, with due authority of Quinnox Consultancy Services. Unauthorized reading, reproduction, publication, use, dissemination, forwarding, printing or copying of this e-mail and its attachments is prohibited. We have checked this message for any known viruses; however we decline any liability, in case of any damage caused by a non-detected virus. For more details about our company, visit http://www.Quinnox.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SNMP monitoring
Hi All, I've been reading up on 1.4 snmp integration. When I try and compile asterisk with a -with-netsnmp option it complains about net-snmp installation being broken. However, the net-snmp-devel rpm is installed, and snmpd on the machine runs fine. Anyone have a guide for the pre-requisites needed ? Cheers, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK -999 dialing issue
On Thursday 14 February 2008 07:55:08 SIP wrote:
Gordon Henderson wrote:
On Thu, 14 Feb 2008, Phil Knighton wrote:
[softoption-zap]
exten = _0[123456789].,1,NoOp(${EXTEN})
exten = _0[123456789].,2,Dial(Zap/g0/${EXTEN},,j)
exten = _0[123456789].,103,Dial(IAX2/Gradwell/44${EXTEN:1},,)
exten = _00[1-9].,1,Dial(IAX2/Gradwell/${EXTEN:2},,)
exten = _90[123456789].,1,Dial(IAX2/Gradwell/44${EXTEN:2},,)
OMG!!!
You're selecting 2 different output channels depending on the number
dialled!!!
(UK or international)...
That's ... LCR!!!
In ... Dialplan!!!
And according to a recent thread, that's like ... impossible, not
recommended, really really hard, with databases and external hardware
required, etc. (!!!)
Not impossible. I think the explanation was that it was ugly. And...
well... that is. Now, imagine sorting through a list of 500,000
possible dialing prefixes (something we have) instead of 3 or 4. Tell me
that would be clean and pretty without a DB lookup.
Anyone can LCR 2 routes in a dialplan, but that's hardly an effective
example of LCR.
Right, and as soon as you add func_odbc to the mix, it becomes easy to query
such a database in the dialplan.
--
Tilghman
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Re: [asterisk-users] SIP over TCP
On 13/02/2008, Raj Jain [EMAIL PROTECTED] wrote: SIP over TCP is included in 1.6. http://svn.digium.com/view/asterisk/tags/1.6.0-beta1/CHANGES?view=co Thanks all! :o) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Burnouts
On Thu, 14 Feb 2008, Brent Davidson wrote: That 3 foot cable run passes behind a 21 monitor that was connected to the server. When the line tests showed everything OK, I decided the monitor might be a long shot but I could understand how the degaussing coil coil could possibly induce a surge on the phone line if the monitor was somehow degaussing nightly, so I unplugged the monitor's power cable and left everything else as it was. So far so good. X100p #4 is still working this morning, so it looks like the problem is solved. Hope this helps someone else later on. Indicative, but not conclusive. It would be interesting to disconnect the pairs from the x100p's and plug them into an oscilloscope or something similar and power-cycle, auto-tune, sleep-mode, and degauss the tube to see if your assumption is reasonable. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Variable setting in AMI Originate
Working with asterisk 1.4; using the AMI Originate command, it is possible to do something like: Variable: CDR(accountcode)123456 Or must the variable names be var[n] where n is a number? I'd like to set the accountcode for a Local channel that originates a call. Thanks. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telephone line signaling configuration in Egypt for FXO ports
Hi; Sorry, I forgot to post the zapata version, it is 1.4 but I do not know the release and I do not know how to know the exact release. Regards Bilal -- Hi All; I am facing a problem that the telephon line in Egypt does not work with the FXO port at the digium card (TDM22B), and I tried to play in loadzone and defaultzone without any success, when we call to the PBX it gives Busy signal sometimes, and othertimes it rings without any response in Asterisk. Is there any other configuration I have to do it to resolve this issue? Any advise about a troubleshooting method to resolve it? What version of zaptel? What do you have in zaptel.conf? Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Emagen (a Telrad VM solution) -- any way to replace with *?
Hi, all. I've got a PoS Emagen VM system tied in with our Telrad PBX. I hate 'em both, but I'm stuck with the Telrad for the time being. That being said, does anyone know of a way to replace the VM solution with Asterisk? I'd -love- to get an Asterisk box in the loop, here. Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telephone line signaling configuration in Egypt for FXO ports
Hi; The PBX located in Egypt at Cairo city. I am able to receive calls on the FXO ports at 3rd and 4th ports, but I am not able to place outgoing call (it gives busy tone that coming from the service provider, or it gives an voice message from the service provider that the dialed number is wrong). I am posting now the needed data that I was able to collect it: Asterisk version: Asterisk SVN-branch-1.4-r90231 localhost*CLI show globals ignorepat=9 TRUNKMSD=1 IAXINFO=guest CONSOLE=Console/dsp PSTNTRUNK=Zap/g1 TRUNK=Zap/g2 zap show status Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 zap show cadences r1: 125,125,2000,4000 r2: 250,250,500,1000,250,250,500,4000 r3: 125,125,125,125,125,4000 r4: 1000,500,2500,5000 zapata.conf: [channels] rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=no hidecallerid=no callwaiting=yes usecallingpres=no callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=no relaxdtmf=yes rxgain=15.0 txgain=15.0 group=1 channel = 3 callgroup=1 pickupgroup=1 immediate=yes busydetect=yes busycount=3 hanguponpolarityswitch=yes callprogress=yes context=Internal signalling=fxo_ks channel = 1,2 context=External signalling=fxs_ks channel = 3,4 zaptel.conf: loadzone = uk defaultzone=uk fxoks=1,2 fxsks=3,4 Any advise? Regards Bilal - Hi All; I am facing a problem that the telephon line in Egypt does not work with the FXO port at the digium card (TDM22B), and I tried to play in loadzone and defaultzone without any success, when we call to the PBX it gives Busy signal sometimes, and othertimes it rings without any response in Asterisk. Is there any other configuration I have to do it to resolve this issue? Any advise about a troubleshooting method to resolve it? What version of zaptel? What do you have in zaptel.conf? Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue
Honestly.. this sounds like a telco issue.I understand what the other person is saying about the PRI still being technically up... BUT... if the channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to the next available channel, which they clearly are not doing. On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith [EMAIL PROTECTED] wrote: Hi Tim, Imagine the scenario where we had 10x Asterisk servers, with calls presenting sequentially starting from the first server, then server two, etc. If we took down the first server for maintenance with 'asterisk -rx stop gracefully' we then will block all incoming calls to all servers as our telco will simply relay the BUSY back to the caller. If there are a number of calls on the first server that continue for another 20 minutes, then all inbounds are blocked for that period of time. We are finding at present we have to look at the calls on the server and make a decision if we are busy to simply reboot the server and hence lose calls. Not ideal but then we don't end up blocking our inbounds. What I was hoping to do was find a way to cause the telco to present the call to the next ISDN30 and therefore would allow us to cleanly take down an Asterisk server for maintenance without causing this issue. In a sense to put the ISDN30 into alarm mode while still continuing the active calls. Do you know if this is at all possible, even if we considered patching zaptel to add this functionality or does the telco rely on the entire PRI being in alarm before it presents the call to the next ISDN30 ? This would allow us to run maintenance on our servers during busy periods without causing disruption, and would be an excellent feature. Many thanks, Andrew -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Tim Nelson *Sent:* 13 February 2008 18:12 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Even if * is shutdown, zaptel is still running and your ISDN channels are still technically up. Shutting down zaptel should close the channels and put those circuits into alarm mode. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Andrew Smith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 12:03:51 PM (GMT-0600) America/Chicago Subject: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Hi there, I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN E1s. Basically our telco is presenting calls in order of the ISDNs on our servers. SERVER1=1,2,3,4 SERVER2=5,6,7,8 We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in alarm and the calls will then presented to PRIs 5,6,7,8 on SERVER2. If I have to take SERVER1 offline for maintenance (asterisk -rx shutdown gracefully) any incoming calls receive a BUSY tone. What I would like to know is if there is anyway to get around this and not send a BUSY back to our callers and somehow allow our telco to present calls immediately to SERVER2. Anyone have any ideas or are we stuck with this behaviour until the calls drop to 0 and Asterisk shuts down ? Thanks, Andrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Burnouts
I considered doing just that, but since I didn't have my scope with me and it's an hour's drive away it didn't seem worth it at this point. If we have trouble again I may take the scope down there and test it. -Brent Steve Edwards wrote: On Thu, 14 Feb 2008, Brent Davidson wrote: That 3 foot cable run passes behind a 21 monitor that was connected to the server. When the line tests showed everything OK, I decided the monitor might be a long shot but I could understand how the degaussing coil coil could possibly induce a surge on the phone line if the monitor was somehow degaussing nightly, so I unplugged the monitor's power cable and left everything else as it was. So far so good. X100p #4 is still working this morning, so it looks like the problem is solved. Hope this helps someone else later on. Indicative, but not conclusive. It would be interesting to disconnect the pairs from the x100p's and plug them into an oscilloscope or something similar and power-cycle, auto-tune, sleep-mode, and degauss the tube to see if your assumption is reasonable. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Telephone line signaling configuration in Egypt for FXO ports
bilal ghayyad wrote: [channels] rxgain=15.0 txgain=15.0 Wow! Is this necessary? Is this something you took from a sample config somewhere, or numbers that you arrived at through trial and error? They seem a bit high in my experience, *but* I've never been to Egypt before, and I sure wouldn't be surprised if this was necessary -- just wanted your confirmation ;) Mojo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] X100P Burnouts
Quoting Brent Davidson [EMAIL PROTECTED]: I considered doing just that, but since I didn't have my scope with me and it's an hour's drive away it didn't seem worth it at this point. If we have trouble again I may take the scope down there and test it. unless the cable is in the same spot (relative to coil, ground, other metal), loaded with the same impedance characteristics as an x100p, etc etc you might not measure anything meaningful anyway. you'd have to wire the scope across it somehow in place without destroying the experiment with the scope leads themselves adding another end of line leg that is not the x100p. I would say you are probably correct in the assumption, and everyone should just bear this sort of thing in mind. if in doubt its easy to plug in through a power bar with phone surge supressor close to the end of the run, or one of the many other devices that do the same thing. -Brent Steve Edwards wrote: On Thu, 14 Feb 2008, Brent Davidson wrote: That 3 foot cable run passes behind a 21 monitor that was connected to the server. When the line tests showed everything OK, I decided the monitor might be a long shot but I could understand how the degaussing coil coil could possibly induce a surge on the phone line if the monitor was somehow degaussing nightly, so I unplugged the monitor's power cable and left everything else as it was. So far so good. X100p #4 is still working this morning, so it looks like the problem is solved. Hope this helps someone else later on. Indicative, but not conclusive. It would be interesting to disconnect the pairs from the x100p's and plug them into an oscilloscope or something similar and power-cycle, auto-tune, sleep-mode, and degauss the tube to see if your assumption is reasonable. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] message: !! Got Busy in Connected State !?!
--- Fons van der Beek [EMAIL PROTECTED] wrote: What phone do you use? Linksys ? SIP softphones and Alcatel analog phones behind ATA gateways (Grandstream). However, I'm having a hard time reproducing the problem. It doesn't happen often. Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue
That does sound like what is happening.. Telco knows channel 1-23 are not busy (so far as they are concerned), however.. so far as you are concerned, they are busy.. so telco sends the call down... but the equipment doesn't take it. I would *think* the Telco could keep trying channels down the hunt group, but maybe not? We have, in the past, seen this issue with our dial-up modem banks.. especially if I would take one offline. However, it is not a big enough issue (i.e. we don't take things down that often) for me to look into it fully. On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly [EMAIL PROTECTED] wrote: I think the problem is that the telco presents the call on a specific channel, then zaptel tells it that the channel is busy. We need to be able to tell the telco that each unused channel on a given span is unavailable, and it will determine that the others are in use and will present the call on a channel on another span. A rather ugly work-around (since Andrew seems to have lots of channels available, and one would assume that maintenance of this nature would occur during slow periods) would be to make calls to a DID in the same trunk group on all idle channels on the span shutting down then, when all channels on the span are in use and none of them are doing anything useful, take the span down hard so the telco will divert all calls to another span. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Matt *Sent:* Thursday, February 14, 2008 2:28 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue Honestly.. this sounds like a telco issue.I understand what the other person is saying about the PRI still being technically up... BUT... if the channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to the next available channel, which they clearly are not doing. On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith [EMAIL PROTECTED] wrote: Hi Tim, Imagine the scenario where we had 10x Asterisk servers, with calls presenting sequentially starting from the first server, then server two, etc. If we took down the first server for maintenance with 'asterisk -rx stop gracefully' we then will block all incoming calls to all servers as our telco will simply relay the BUSY back to the caller. If there are a number of calls on the first server that continue for another 20 minutes, then all inbounds are blocked for that period of time. We are finding at present we have to look at the calls on the server and make a decision if we are busy to simply reboot the server and hence lose calls. Not ideal but then we don't end up blocking our inbounds. What I was hoping to do was find a way to cause the telco to present the call to the next ISDN30 and therefore would allow us to cleanly take down an Asterisk server for maintenance without causing this issue. In a sense to put the ISDN30 into alarm mode while still continuing the active calls. Do you know if this is at all possible, even if we considered patching zaptel to add this functionality or does the telco rely on the entire PRI being in alarm before it presents the call to the next ISDN30 ? This would allow us to run maintenance on our servers during busy periods without causing disruption, and would be an excellent feature. Many thanks, Andrew -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Tim Nelson *Sent:* 13 February 2008 18:12 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Even if * is shutdown, zaptel is still running and your ISDN channels are still technically up. Shutting down zaptel should close the channels and put those circuits into alarm mode. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Andrew Smith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 12:03:51 PM (GMT-0600) America/Chicago Subject: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Hi there, I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN E1s. Basically our telco is presenting calls in order of the ISDNs on our servers. SERVER1=1,2,3,4 SERVER2=5,6,7,8 We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in alarm and the calls will then presented to PRIs 5,6,7,8 on SERVER2. If I have to take SERVER1 offline for maintenance (asterisk -rx shutdown gracefully) any incoming calls receive a BUSY tone. What
Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue
I think the problem is that the telco presents the call on a specific channel, then zaptel tells it that the channel is busy. We need to be able to tell the telco that each unused channel on a given span is unavailable, and it will determine that the others are in use and will present the call on a channel on another span. A rather ugly work-around (since Andrew seems to have lots of channels available, and one would assume that maintenance of this nature would occur during slow periods) would be to make calls to a DID in the same trunk group on all idle channels on the span shutting down then, when all channels on the span are in use and none of them are doing anything useful, take the span down hard so the telco will divert all calls to another span. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, February 14, 2008 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue Honestly.. this sounds like a telco issue.I understand what the other person is saying about the PRI still being technically up... BUT... if the channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to the next available channel, which they clearly are not doing. On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith [EMAIL PROTECTED] wrote: Hi Tim, Imagine the scenario where we had 10x Asterisk servers, with calls presenting sequentially starting from the first server, then server two, etc. If we took down the first server for maintenance with 'asterisk -rx stop gracefully' we then will block all incoming calls to all servers as our telco will simply relay the BUSY back to the caller. If there are a number of calls on the first server that continue for another 20 minutes, then all inbounds are blocked for that period of time. We are finding at present we have to look at the calls on the server and make a decision if we are busy to simply reboot the server and hence lose calls. Not ideal but then we don't end up blocking our inbounds. What I was hoping to do was find a way to cause the telco to present the call to the next ISDN30 and therefore would allow us to cleanly take down an Asterisk server for maintenance without causing this issue. In a sense to put the ISDN30 into alarm mode while still continuing the active calls. Do you know if this is at all possible, even if we considered patching zaptel to add this functionality or does the telco rely on the entire PRI being in alarm before it presents the call to the next ISDN30 ? This would allow us to run maintenance on our servers during busy periods without causing disruption, and would be an excellent feature. Many thanks, Andrew _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Nelson Sent: 13 February 2008 18:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Even if * is shutdown, zaptel is still running and your ISDN channels are still technically up. Shutting down zaptel should close the channels and put those circuits into alarm mode. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Andrew Smith [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, February 13, 2008 12:03:51 PM (GMT-0600) America/Chicago Subject: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Hi there, I currently have multiple Asterisk servers using Sangoma A104d Quad ISDN E1s. Basically our telco is presenting calls in order of the ISDNs on our servers. SERVER1=1,2,3,4 SERVER2=5,6,7,8 We have redundancy in that if SERVER1 is shutdown then each ISDN PRI is in alarm and the calls will then presented to PRIs 5,6,7,8 on SERVER2. If I have to take SERVER1 offline for maintenance (asterisk -rx shutdown gracefully) any incoming calls receive a BUSY tone. What I would like to know is if there is anyway to get around this and not send a BUSY back to our callers and somehow allow our telco to present calls immediately to SERVER2. Anyone have any ideas or are we stuck with this behaviour until the calls drop to 0 and Asterisk shuts down ? Thanks, Andrew ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
[asterisk-users] ExtenSpy strange behavior on Asterisk 1.4.18
Hi list,
I have been experiencing a strange behavior with asterisk and i would
like to know if someone else has face it.
This is my scenario,
3 extensions created on sip.conf: 121 | 123 | 123
Everything work just perfect except for the following issue:
I have this block on my extensions.conf
[record] ;---Extensiones individuales
exten = _7781[23]X,1,Authenticate(/etc/asterisk/eavepass|am)
; Authenticate
exten =
_7781[23]X,n,extenspy(${EXTEN:3:[EMAIL PROTECTED]|bqr(${EXTEN:3:6}-)) ;
extenspy([EMAIL PROTECTED]|bqr(fileprefix-)
exten = _7781[23]X,n,Hangup
; Hangup
If 122 calls 123 , and 123 answers ( 122 =[called] 123 ):
- Dialing 778*122* from 121 (for spying and recording) generates a 0
byte file like 122-.1203025287.raw (on /var/spool/asterisk/monitor/)
- Dialing 778*123* from 121 (for spying and recording) generates a
byte-significant file like 123-.1203025598.raw
If 123 calls 122 , and 122 answers ( 123 =[called] 122 ):
- Dialing 778*122* from 121 (for spying and recording) generates a
byte-significant file like 122-.1203025808.raw
- Dialing 778*123* from 121 (for spying and recording) generates a 0
byte file like 123-.1203025923.raw
In the situation where the 0 byte file is generated, NO AUDIO is
listened on 121; I have NO IDEA of what could be causing this, as
there's no apprently refference on the documentation. Could this be a
possible bug?
Regards,
# --
# Jose P. Espinal
# http://www.slackware-es.com
# --
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Re: [asterisk-users] SNMP monitoring
I had the same problem some time ago... You got to install also this packages: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel That should do it! Regards, Ricardo Carvalho. On Thu, Feb 14, 2008 at 5:30 PM, Adrian Marsh [EMAIL PROTECTED] wrote: Hi All, I've been reading up on 1.4 snmp integration. When I try and compile asterisk with a –with-netsnmp option it complains about net-snmp installation being broken. However, the net-snmp-devel rpm is installed, and snmpd on the machine runs fine. Anyone have a guide for the pre-requisites needed ? Cheers, Adrian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK -999 dialing issue
Steve Langstaff [EMAIL PROTECTED] writes: Oops! Yes, I see that now - my fault for confusing Asterisk pattern matching with RFC3435 pattern matching. Sorry. Unfortunately inventing a new regex syntax seems to be a favourite pastime. Perhaps it would be possible to allow exten = /00.*/,Dial... It might cause problems with the ex-GF syntax. Another starting character could mean RFC3435 pattern matching. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor Asterisk
Matthew J. Roth [EMAIL PROTECTED] writes: Yes, asterisk -rx will only allow you to execute CLI commands. It also tends to spew out a bunch of garbage that makes parsing difficult unless verbosity is always set to 0. It would be very handy if it was possible to turn off messages that aren't the direct result of a command in a particular instance of asterisk -r. Perhaps asterisk -r -q? /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK -999 dialing issue
On Thu, 2008-02-14 at 22:32 +0100, Benny Amorsen wrote:
Perhaps it would be possible to allow exten = /00.*/,Dial... It might
cause problems with the ex-GF syntax. Another starting character could
mean RFC3435 pattern matching.
I've been suggesting that for about four years now (long before I ever
started working for Digium), but the core Asterisk developers tell me it
will have a very negative impact on Asterisk performance. I'd obviously
like to see it use a different character to inform the dialplan parser
that we're using a different pattern matching system, so that we can
limit the performance impact to just those extensions that require it.
But, for now, I've lived to learn to get along with the things that the
dialplan provides. Don't forget that we have two different regex
operators what we can use inside of an Asterisk dialplan expression. :-)
exten = _X.,1,GotoIf($[${EXTEN} : /#+[2-7][0-9]{3}/]?happy)
--
Jared Smith
Community Relations Manager
Digium, Inc.
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[asterisk-users] gtalk and dtmf
Hi, i've just finished setting up gtalk connection with asterisk. it works nice, audio is full duplex. i just have one question which i could not find an exact answer to. Is gtalk able to send dtmf codes? Because i'd like to listen to my voicemails while away from home. I've been googling for half an hour, i found some sort of jingle protocol which i'm not sure what to use for but it might be the solution? It seems to me that my problem is sending the dtmf tones, not receiving them, so this is really gtalk related. I'm writing here because i read many of you have successfully integrated gtalk to asterisk and hoping somebody have a solution or at least some direction where i can move forward to. thanks Adam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restart asterisk daily
If you want to flush your disk cache to see how much memory is being eaten cache pages, try this: echo 3 /proc/sys/vm/drop_caches - ast erisk [EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue
If you take Asterisk down, the PRI should go down as the D channel is down. Then the telco should KNOW that there is trouble with the PRI and those channels are in trouble busy and not availible. If the telco still tries to push a call to a channel on a PRI that is down, then the telco is at fault. Lyle Matt wrote: That does sound like what is happening.. Telco knows channel 1-23 are not busy (so far as they are concerned), however.. so far as you are concerned, they are busy.. so telco sends the call down... but the equipment doesn't take it. I would *think* the Telco could keep trying channels down the hunt group, but maybe not? We have, in the past, seen this issue with our dial-up modem banks.. especially if I would take one offline. However, it is not a big enough issue (i.e. we don't take things down that often) for me to look into it fully. On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I think the problem is that the telco presents the call on a specific channel, then zaptel tells it that the channel is busy. We need to be able to tell the telco that each unused channel on a given span is unavailable, and it will determine that the others are in use and will present the call on a channel on another span. A rather ugly work-around (since Andrew seems to have lots of channels available, and one would assume that maintenance of this nature would occur during slow periods) would be to make calls to a DID in the same trunk group on all idle channels on the span shutting down then, when all channels on the span are in use and none of them are doing anything useful, take the span down hard so the telco will divert all calls to another span. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *On Behalf Of *Matt *Sent:* Thursday, February 14, 2008 2:28 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue Honestly.. this sounds like a telco issue.I understand what the other person is saying about the PRI still being technically up... BUT... if the channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to the next available channel, which they clearly are not doing. On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Tim, Imagine the scenario where we had 10x Asterisk servers, with calls presenting sequentially starting from the first server, then server two, etc. If we took down the first server for maintenance with 'asterisk -rx stop gracefully' we then will block all incoming calls to all servers as our telco will simply relay the BUSY back to the caller. If there are a number of calls on the first server that continue for another 20 minutes, then all inbounds are blocked for that period of time. We are finding at present we have to look at the calls on the server and make a decision if we are busy to simply reboot the server and hence lose calls. Not ideal but then we don't end up blocking our inbounds. What I was hoping to do was find a way to cause the telco to present the call to the next ISDN30 and therefore would allow us to cleanly take down an Asterisk server for maintenance without causing this issue. In a sense to put the ISDN30 into alarm mode while still continuing the active calls. Do you know if this is at all possible, even if we considered patching zaptel to add this functionality or does the telco rely on the entire PRI being in alarm before it presents the call to the next ISDN30 ? This would allow us to run maintenance on our servers during busy periods without causing disruption, and would be an excellent feature. Many thanks, Andrew *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *On Behalf Of *Tim Nelson *Sent:* 13 February 2008 18:12 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Even if * is shutdown, zaptel is still running and your ISDN
Re: [asterisk-users] SNMP monitoring
Ricardo Carvalho wrote: I had the same problem some time ago... You got to install also this packages: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel That should do it! Why would this depend on newt? net-snmp and lm-sensor headers and libraries make sense. newt doesn't make any sense as a dependency. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue
Correct me if I'm wrong, but as I understand it your issue is that when you give Asterisk the stop gracefully command it waits until all active calls have finished before it takes the ISDN down but gives busy signals to new incoming calls on idle channels. If this is the case then it would seem that Asterisk is actually answering the call on the incoming channel and playing a busy signal. From reading a couple of threads on another list it appears this is the case (Google: Asterisk busy out PRI to find the discussion). There also appears to be some interest in making a function do what you need in the future. For the time being, however, a simple solution would be to create a temporary dial-plan that follows each outgoing hangup with a dial command to either a test number or some other service that will just keep playing audio down the line and not hangup. (You'd probably need to set some variable to know which channels had been busied) When you need to take down a server, load this dial plan and wait for all channels to call the busy number, then hang them all up and issue a stop now. It's a messy solution, but it's all I can think of without hacking code. The only other way I'd know would be to hack the code for the dial or answer command and build another command that simply takes the channel off-hook and leaves it there. Good luck, Brent Davidson Lyle Giese wrote: If you take Asterisk down, the PRI should go down as the D channel is down. Then the telco should KNOW that there is trouble with the PRI and those channels are in trouble busy and not availible. If the telco still tries to push a call to a channel on a PRI that is down, then the telco is at fault. Lyle Matt wrote: That does sound like what is happening.. Telco knows channel 1-23 are not busy (so far as they are concerned), however.. so far as you are concerned, they are busy.. so telco sends the call down... but the equipment doesn't take it. I would *think* the Telco could keep trying channels down the hunt group, but maybe not? We have, in the past, seen this issue with our dial-up modem banks.. especially if I would take one offline. However, it is not a big enough issue (i.e. we don't take things down that often) for me to look into it fully. On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I think the problem is that the telco presents the call on a specific channel, then zaptel tells it that the channel is busy. We need to be able to tell the telco that each unused channel on a given span is unavailable, and it will determine that the others are in use and will present the call on a channel on another span. A rather ugly work-around (since Andrew seems to have lots of channels available, and one would assume that maintenance of this nature would occur during slow periods) would be to make calls to a DID in the same trunk group on all idle channels on the span shutting down then, when all channels on the span are in use and none of them are doing anything useful, take the span down hard so the telco will divert all calls to another span. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax *From:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] *On Behalf Of *Matt *Sent:* Thursday, February 14, 2008 2:28 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue Honestly.. this sounds like a telco issue.I understand what the other person is saying about the PRI still being technically up... BUT... if the channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to the next available channel, which they clearly are not doing. On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi Tim, Imagine the scenario where we had 10x Asterisk servers, with calls presenting sequentially starting from the first server, then server two, etc. If we took down the first server for maintenance with 'asterisk -rx stop gracefully' we then will block all incoming calls to all servers as our telco will simply relay the BUSY back to the caller. If there are a number of calls on the first server that continue for another 20 minutes, then all inbounds are blocked for that period of time. We are finding at present we have to look at the calls on the server and make a decision if we are busy to simply reboot the server and hence lose calls. Not ideal but then we don't end up
Re: [asterisk-users] SNMP monitoring
Maybe you'r right and newt isn't really necessary. I just read somewhere that those dependencies were needed, I've installed them and it worked... Try to only install the other ones and if res_snmp gets compiled without it, great! Regards, Ricardo Carvalho. On Fri, Feb 15, 2008 at 12:01 AM, Darrick Hartman (lists) [EMAIL PROTECTED] wrote: Ricardo Carvalho wrote: I had the same problem some time ago... You got to install also this packages: net-snmp-devel newt-devel lm_sensors-devel bzip2-devel That should do it! Why would this depend on newt? net-snmp and lm-sensor headers and libraries make sense. newt doesn't make any sense as a dependency. Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue
Andrew wants to take the system down softly-there are active calls on some channels. He doesn't want to accept additional calls on the idle channels. He can't take the D channel down without disruption to the active calls. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese Sent: Thursday, February 14, 2008 5:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue If you take Asterisk down, the PRI should go down as the D channel is down. Then the telco should KNOW that there is trouble with the PRI and those channels are in trouble busy and not availible. If the telco still tries to push a call to a channel on a PRI that is down, then the telco is at fault. Lyle Matt wrote: That does sound like what is happening.. Telco knows channel 1-23 are not busy (so far as they are concerned), however.. so far as you are concerned, they are busy.. so telco sends the call down... but the equipment doesn't take it. I would *think* the Telco could keep trying channels down the hunt group, but maybe not? We have, in the past, seen this issue with our dial-up modem banks.. especially if I would take one offline. However, it is not a big enough issue (i.e. we don't take things down that often) for me to look into it fully. On Thu, Feb 14, 2008 at 4:07 PM, Don Kelly [EMAIL PROTECTED] wrote: I think the problem is that the telco presents the call on a specific channel, then zaptel tells it that the channel is busy. We need to be able to tell the telco that each unused channel on a given span is unavailable, and it will determine that the others are in use and will present the call on a channel on another span. A rather ugly work-around (since Andrew seems to have lots of channels available, and one would assume that maintenance of this nature would occur during slow periods) would be to make calls to a DID in the same trunk group on all idle channels on the span shutting down then, when all channels on the span are in use and none of them are doing anything useful, take the span down hard so the telco will divert all calls to another span. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Thursday, February 14, 2008 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN PRIs and taking a server down formaintenance - blocking issue Honestly.. this sounds like a telco issue.I understand what the other person is saying about the PRI still being technically up... BUT... if the channel is BUSY/BLOCKED/WHATEVER, the Telco should be forwarding the call to the next available channel, which they clearly are not doing. On Thu, Feb 14, 2008 at 8:29 AM, Andrew Smith [EMAIL PROTECTED] wrote: Hi Tim, Imagine the scenario where we had 10x Asterisk servers, with calls presenting sequentially starting from the first server, then server two, etc. If we took down the first server for maintenance with 'asterisk -rx stop gracefully' we then will block all incoming calls to all servers as our telco will simply relay the BUSY back to the caller. If there are a number of calls on the first server that continue for another 20 minutes, then all inbounds are blocked for that period of time. We are finding at present we have to look at the calls on the server and make a decision if we are busy to simply reboot the server and hence lose calls. Not ideal but then we don't end up blocking our inbounds. What I was hoping to do was find a way to cause the telco to present the call to the next ISDN30 and therefore would allow us to cleanly take down an Asterisk server for maintenance without causing this issue. In a sense to put the ISDN30 into alarm mode while still continuing the active calls. Do you know if this is at all possible, even if we considered patching zaptel to add this functionality or does the telco rely on the entire PRI being in alarm before it presents the call to the next ISDN30 ? This would allow us to run maintenance on our servers during busy periods without causing disruption, and would be an excellent feature. Many thanks, Andrew _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Nelson Sent: 13 February 2008 18:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ISDN PRIs and taking a server down for maintenance - blocking issue Even if * is shutdown, zaptel is still running and your ISDN channels are still technically up. Shutting down zaptel should close the channels and put those circuits into alarm mode. Tim Nelson
Re: [asterisk-users] R: GXP2000 and asterisk 1.0.9
On Thu, Feb 14, 2008 at 10:12 AM, Henry Devito [EMAIL PROTECTED] wrote: I had GXP-2000's running on 1.0 versions of asterisk even earlier. So I know it does work. I upgraded one of my customers GXP's to the latest I'm not sure you are right, since I have had Polycoms that didn't work, it's quite possible you should have GPXs that do work. firmware in it still works. Can you post the output of the CLI with verbose set to 99 and the the output from the asterisk log file that has the call in it. You can usually do a 'tail /var/log/asterisk/full -n 400' right after the call fails. I will be glad to help, just need a little more info to narrow down the issue. Thanks Henry - Original Message - From: Giordano Grandis [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 14, 2008 2:15 AM Subject: [asterisk-users] R: GXP2000 and asterisk 1.0.9 1. The phone has not the DND active, i checked it several times 2. Outbound calls always success, the problem is when the phone receive a call, it repsnds with busy signalling. 3. The firmware i just the lastest one 1.1.5.15 and i cannot upgrade asterisk. Thanks for all -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di C F Inviato: mercoledì 13 febbraio 2008 21.09 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] GXP2000 and asterisk 1.0.9 Just check DND if it's on on the phone or not. What is the CLI output when you try making a phone call? Why don't you try it with a later version of astrisk and a Phone? On Feb 13, 2008 10:58 AM, Giordano Grandis [EMAIL PROTECTED] wrote: Hi all gusy, i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few go in busy state, if you call it get the busy tone but the phone can male any type of call. This is my sip.conf [502] language = it username = 502 secret = password host = dynamic type = friend context = local canreinvite = yes dtmfmode = info callgroup = 1 pickupgroup = 1 callerid = 502 502 Under Grandstream's support suggest, I set Use randmom port to yes and Nat traversal (STUN) to No, but send keep alive but without success. This is the firmware version: Program-- 1.1.5.15Bootloader-- 1.1.5.6 Anyone can help me ? Thanks in advance Giordano No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 15.20 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.20.4/1277 - Release Date: 13/02/2008 20.00 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DialPlan help with Analog Fax Machine
I'm struggling to get my dialplan to work with a simple analog fax
machine.
I have TDM400B zaptel card with an FXO and FXS port. I have the FXO
port connected to the POTS machine and the FAX machine connected to the
FXS port.
The FAX machine itself works fine, I can FAX outgoing messages fine. I
can also dial the FAX extension from the internal context, the FAX
machine answers and I hear the FAX tones.
I'm struggling to get the fax detection to work, causing a transfer to
the FAX machine. I think the fax transfer starts, but for some reason
the dialplan falls through and the connection is dropped immediately.
This should be so simple ...
version:
asterisk*CLI core show version
Asterisk 1.6.0-beta2 built by jduda @ asterisk on a i686 running Linux
on 2008-02-03 03:23:54 UTC
zapata.conf has:
; FAX machine connected here
;immediate=no
;busydetect=yes
;busycount=8
;musiconhold=default
faxdetect=no
signalling=fxo_ks
context=internal
channel = 1
; PSTN connected here
;immediate=no
;busydetect=yes
;busycount=8
;musiconhold=default
mwimonitor=yes
;mwilevel=512
mwimonitornotify=/usr/local/sbin/zapnotify.sh
faxdetect=incoming
signalling=fxs_ks
context=incoming
channel = 4
extensions.conf has:
[fax-hardware]
exten = s,1,StopPlaytones
exten = s,2,Dial(ZAP/1,40,tr)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[incoming]
exten = fax,1,Goto(fax-hardware,s,1)
exten = s,1,Goto(incoming-dial,s,1)
exten = my phone number,1,Goto(incoming-dial,s,1)
[incoming-dial]
exten = s,1,Zapateller(nocallerid)
exten = s,2,SetMusicOnHold(icecast)
exten =
s,3,GotoIf(${DB_EXISTS(blacklist/${CALLERID(number)})}?custom-blacklisted,s,1)
exten = s,4,Set(DB(CALLTRACE/lastcaller)=${CALLERID(number)})
exten = s,5,AGI(MisterHouse.agi,CallerID)
exten = s,6,Answer
exten = s,7,Playtones(ring)
exten =
s,8,Dial(${PHONES0}${PHONES1}${PHONES2}${PHONES7}${PHONES11},20,tr)
exten = s,9,Goto(s-${DIALSTATUS},1) ; if no fax, branch on dialstatus
exten = s-NOANSWER,1,Macro(voicemail,${PHONES0VM})
exten = s-NOANSWER,2,Hangup()
exten = s-BUSY,1,Macro(voicemail,${PHONES0VM})
exten = s-BUSY,2,Hangup()
exten = _s-.,1,Goto(s-NOANSWER,1) ; everything else is treated as no
answer
exten = s,105,Goto(5)
exten = fax,1,Goto(fax-hardware,s,1)
When the FAX call comes in, I get this:
[Feb 14 20:01:03] NOTICE[1826]: chan_zap.c:7306 mwi_thread: Got event 18
(Ring Begin)... Passing along to ss_thread
-- Starting simple switch on 'Zap/4-1'
[Feb 14 20:01:04] NOTICE[1826]: chan_zap.c:7066 ss_thread: Got event 2
(Ring/Answered)...
-- Executing [EMAIL PROTECTED]:1] Goto(Zap/4-1, incoming-dial,s,1) in
new stack
-- Goto (incoming-dial,s,1)
-- Executing [EMAIL PROTECTED]:1] Zapateller(Zap/4-1,
nocallerid) in new stack
-- Executing [EMAIL PROTECTED]:2] SetMusicOnHold(Zap/4-1,
icecast) in new stack
-- Executing [EMAIL PROTECTED]:3] GotoIf(Zap/4-1,
0?custom-blacklisted,s,1) in new stack
-- Executing [EMAIL PROTECTED]:4] Set(Zap/4-1,
DB(CALLTRACE/lastcaller)=8884732963) in new stack
-- Executing [EMAIL PROTECTED]:5] AGI(Zap/4-1,
MisterHouse.agi,CallerID) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/MisterHouse.agi
MisterHouse.agi,CallerID: AGI Environment Dump:
MisterHouse.agi,CallerID: -- accountcode =
MisterHouse.agi,CallerID: -- arg_1 = CallerID
MisterHouse.agi,CallerID: -- callerid = 8884732963
MisterHouse.agi,CallerID: -- calleridname = UNAVAILABLE
MisterHouse.agi,CallerID: -- callingani2 = 0
MisterHouse.agi,CallerID: -- callingpres = 0
MisterHouse.agi,CallerID: -- callingtns = 0
MisterHouse.agi,CallerID: -- callington = 0
MisterHouse.agi,CallerID: -- channel = Zap/4-1
MisterHouse.agi,CallerID: -- context = incoming-dial
MisterHouse.agi,CallerID: -- dnid = unknown
MisterHouse.agi,CallerID: -- enhanced = 0.0
MisterHouse.agi,CallerID: -- extension = s
MisterHouse.agi,CallerID: -- language = en
MisterHouse.agi,CallerID: -- priority = 5
MisterHouse.agi,CallerID: -- rdnis = unknown
MisterHouse.agi,CallerID: -- request = MisterHouse.agi
MisterHouse.agi,CallerID: -- threadid = -1232077936
MisterHouse.agi,CallerID: -- type = Zap
MisterHouse.agi,CallerID: -- uniqueid = 1203037263.20
MisterHouse.agi,CallerID: -- version = 1.6.0-beta2
MisterHouse.agi,CallerID: here CallerID
MisterHouse.agi,CallerID: CallerID: 8884732963 Line: Zap/4-1
-- Zap/4-1AGI Script MisterHouse.agi completed, returning 0
-- Executing [EMAIL PROTECTED]:6] Answer(Zap/4-1, ) in new stack
-- Executing [EMAIL PROTECTED]:7] PlayTones(Zap/4-1, ring) in
new stack
-- Executing [EMAIL PROTECTED]:8] Dial(Zap/4-1,
SIP/100SIP/101SIP/102SIP/107SIP/111,20,tr) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 100
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
[asterisk-users] 57iCT BLF problem
We upgrade 2 of our Aastra 57iCT to the latest firmware (2.1.2.30) and the BLF indicators no longer function. Has anyone had a similar issue? And a solution? PaulH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HPEC
Just wondering how your experience is with HPEC, Is it just for analog interfaces or we can use it on TE122 as well? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] patch which makes Asterisk-Addons 1.4.5 work when codec negotiation patch applied to asterisk
Hi, Since the original codec negotiation patch ( http://bugs.digium.com/view.php?id=4825 report) just closed yesterday, and as well as my report (http://bugs.digium.com/view.php?id=11998), I had nothing to do but send my patches to the list. It might be good if my patches are placed at http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch, but don't know whom should I contact. Anyway sending here. thanks, Ganbold chan_h323.c.patch1 Description: Binary data ooh323cDriver.c.patch Description: Binary data ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restart asterisk daily
Always rely on free -m to see how much free memory you have not top. in terms of memory leak, i have asterisk running on servers with uptime of 400 days (CentOs), if there was any leak, i'm guessing i would have crashed server long time ago. On Thu, Feb 14, 2008 at 4:23 PM, Doug Bailey [EMAIL PROTECTED] wrote: If you want to flush your disk cache to see how much memory is being eaten cache pages, try this: echo 3 /proc/sys/vm/drop_caches - ast erisk [EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restart asterisk daily
On Thu, Feb 14, 2008 at 8:38 PM, Al lists [EMAIL PROTECTED] wrote: Always rely on free -m to see how much free memory you have not top. You could install and use htop - it's a much more functional (and informative) version of top. It shows the difference between shared/buffer/cache memory. -erik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restart asterisk daily
Al lists wrote: Always rely on free -m to see how much free memory you have not top. in terms of memory leak, i have asterisk running on servers with uptime of 400 days (CentOs), if there was any leak, i'm guessing i would have crashed server long time ago. On Thu, Feb 14, 2008 at 4:23 PM, Doug Bailey [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If you want to flush your disk cache to see how much memory is being eaten cache pages, try this: echo 3 /proc/sys/vm/drop_caches - ast erisk [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You will see asterisk behave its worst with multiple queues and heavy dialplan logic. I restart my boxes with queues everynight at midnight just to reset the queue stats displayed with show queue. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restart asterisk daily
On Thu, Feb 14, 2008 at 09:32:04PM -0600, Erik Anderson wrote: On Thu, Feb 14, 2008 at 8:38 PM, Al lists [EMAIL PROTECTED] wrote: Always rely on free -m to see how much free memory you have not top. You could install and use htop - it's a much more functional (and informative) version of top. It shows the difference between shared/buffer/cache memory. It also consumes more CPU. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] restart asterisk daily
On Thu, Feb 14, 2008 at 9:37 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote: It also consumes more CPU. True, a fraction more. If you have that little overhead on your server, though, that this would cause a problem, you probably should upgrade your hardware, IMHO. -eriik ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is a secure call?
2008/2/13, Johansson Olle E [EMAIL PROTECTED]: In SIP, there's a specification for how I as a domain owner can request all calls to my domain to use SIP/TLS by using DNS NAPTR and SRV records. Which one ? Does it also deal with SPIT ? But how do I as a caller request a secure service? I think SPIT is a major concern (though I've not heard a single case of SPIT abuse, yet). How do we place a secure call with DIAL? Do we need SECUREDIAL? Any ideas and thoughts on the subject are welcome! Regards, /Olle ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime SIP peers - reloading cached info
2008/2/13, Atis Lezdins [EMAIL PROTECTED]: On 2/13/08, Rob Hillis [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 If it is being removed in 1.6, I'm a little concerned since there's no mention of this when you show the application, nor on voip-info.org . What application/function is it being replaced by? There's an obsolete warning in 1.4.18, but i somehow remember that it's obsolete already since some 1.4.11 It's func_realtime as i said before. usage shouldn't be much different, you can replace with: Set(REALTIME(sip_buddies,name,100,my_field)=foo); Also, seems that func_realtime will soon support SQL INSERT's and DELETE's :) Do you mean it should be added between today's beta and future GA 1.6 ? Regards, Atis Atis Lezdins wrote: | On 2/13/08, Rob Hillis [EMAIL PROTECTED] wrote: | -BEGIN PGP SIGNED MESSAGE- | Hash: SHA1 | | Atis Lezdins wrote: | | By RealTimeUpdate do you mean func_realtime? It shouldn't care, as | | cache is not implemented in realtime level, but higher (chan_sip). | | | | Are you sure you need sip show XXX load. If you sip prune peer | | data, it should be re-loaded on next access. | | | | What i was suggesting - to dig into chan_sip and create dialplan | | application SipPrune(peer) that would prune the peer directly, by | | using corresponding function - sip_prune_peer() in chan_sip.c - that | | way you will gain some extra performance, as there's no manager/cli | | overhead. | | | | However if you're uncomfortable with C, the app_system shouldn't cause | | any troubles :) | | RealTimeUpdate is more likely to correspond to app_realtime rather than | func_realtime. | | As to my knowledge - that is obsolete and being removed in 1.6, | func_realtime replaces it. That's why i wondered about name - I just | never happened to use it :) | | Regards, | Atis | | -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Remi - http://enigmail.mozdev.org iD8DBQFHsnaM6uKn5cBSgGQRAo/TAKDCruPrn2nm2XV/PYbfSuBKA0j5OwCfQ/Ox QE3SYEmZ01QHUT4ITwmLnT0= =SKEW -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to check if a local channel member of a queue?
Hi, I am using asterisk-1.4.15 I have a queue with one agent added using AddQueueMember (FAO|Local/[EMAIL PROTECTED]|0||Agent/602). Once this command executes queue show FAO shows: FAO has 0 calls (max unlimited) in 'roundrobin' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 60s Members: Agent/602 (dynamic) (Not in use) has taken no calls yet There is no mention of the fact that which channel is used by Agent/602. I am adding agents to queue both from an agi using AddQueueMember, via phone, and manager command QueueAdd via web. In both cases I needs to find out whether the given sip channel has logged in to any queue previously, for proper validation. show agents command in previous version gave such details. How can I get the details given out by show agents in the new 1.4 scheme of things? With much regards, raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk DNS SIP issue
The other day my asterisk local SIP clients got hung when my provider had a DNS failure. All registrations went dead (even the ones that were IP addresses) and all sip peers went offline. I know this was know problem at one point is there any update on this when using a FQDN for one of the peer addresses in sip.conf? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
